A(1) Field of the invention
The invention relates to an interpolating time-discrete filter arrangement for converting a time-discrete input signal with which an input sampling frequency f.sub.i is associated into a time-discrete output signal with which an output sampling frequency f.sub.u is associated which is higher than the input sampling frequency.
A(2) Description of the prior art
As is generally known, a time-discrete signal is formed by a series of signal samples. The sampling frequency associated with such a signal indicates the rate at which these signal samples occur. The signal sample itself indicates the magnitude of the signal at a given instant. Within a certain range such a signal sample can assume any value, or only a number of discrete values. In the latter case a digital signal is involved and the signal sample is usually represented by a code word having a plurality of bits.
Hereafter the signal samples of the input signal will be called input samples and be denoted x(q); q= . . . -2, -1, 0, 1, 2, 3, . . . . Similarly, the signal samples of the output signal will be called output samples and be denoted y(n); n= . . . -2, -1, 0, 1, 2, 3, . . . .
Interpolating filter arrangements of the above-mentioned type have been known for many years already. For the sake of brevity, for a general survey reference is made to references 1-6 listed in paragraph C. They produce a time-discrete output signal with which an output sampling frequency is associated with such a value that the ratio between the output sampling frequency and the input sampling frequency is a rational number. Usually the output sampling frequency is an integral multiple of the input sampling frequency.
Practical implementations of interpolating filters are extensively described in, for example, the references 3, 4 and 5. As all types of time-discrete filter arrangements they comprise a signal processing circuit to which the time-discrete input signal and also filter coefficients are applied. As is known, these filter coefficients represent samples of the finite impulse response of the filter and are produced by a filter coefficients generator.
However, in practice it has been found that there are situations in which the output sampling frequency is not a rational multiple of the input sampling frequency; it holds, for example, that f.sub.u =f.sub.i .sqroot.2. Such a situation is found in, for example, digital audio equipment which must be intercoupled; for example a digital tuner, a digital tape recorder, a digital record-player, etc. In practice these apparatuses each comprise their own clock generator for generating the sampling pulses required. The frequencies of these clock pulse generators will never be perfectly equal to each other. So as to enable the apparatuses to cooperate with each other the output sampling frequency associated with the digital signal produced by a first apparatus must be made equal to the input sampling frequency accepted by the second apparatus.